similar to: Line in use

Just starting to enjoy the full features of asterisk, I do have a couple questions though, that I can't seem to find answers for in the wiki, just wondering if someone could light my way. after a caller has made their choice of options in the dial plan, I would like them to be placed on "hold" (music, not ringing) while the system processes through the rest of the dial plan

Hey, I would like in the course of dial plan logic, to trigger a separate outbound call. If that outbound call is answered, and if that certain key response is detected then it will bridge the incoming call to the newly dialed outbound call. What I want to accomplish is that when a caller dials in, they can enter enter an extension that will call out to a callee's cell phone. When the

There is plenty of information on the wiki for setting asterisk up for transferring calls both from the Dail() command, and features.conf. What really seems to be missing, is simply how do you actually perform the transfer? Blind transfers are pretty simple as you only have two obvious steps. How though do you do attended transfers? 1.) You have a call 2.) You dial *2 or whatever you have

I'm having a real problem with one of my linksys pap2. On outgoing calls the callee will ring, but caller (pap2) will not here it ring When the callee answers, no audio is transmitted either way. Asterisk reports the call connected and bridged correctly. Now the kicker is that sometimes it works and other times it doesn't. I have had the most luck calling land lines, but sometime

For multiline phones how do you set SIP channels to busy. For instance if SIP/101 is on a call then dial would return busy. Right now it just starts ringing on line X, and stacks up from there. What would be really great is if I could control how many calls by the context. So if a call was routed via [overload] Then the ext wouldn't report busy it would just keep ringing available

I'm really confused on how to use these two options together: A while back: JustRumours edited this page: http://www.voip-info.org/wiki-Asterisk+cmd+MeetMe and added a little section about dynamic conferences. the 'e' option is repeated all over the page as the savior of dynamic conferences, maybe I'm just dumb, but can someone tell me if a conference is created with the e

Miles Scruggs wrote: >>Miles Scruggs wrote: >> >>>How do I shorten my servername that is displayed to windows clients >>>currently it is >>>'Samba 3.0.2a-Debian (netbios name)' How do I change this to netbios >> >>name >> >>>only? >> >>Look for the "Server string" line in your smb.conf - this is where

So I hacked up the Dial app to return a numeric return code instead of changing contexts based on a number being busy or unanswered. The purpose for this modified dial app, which I call AGIDial, is to help me concoct a "follow-me" type of application. The app returns -1 for a completed call, 0 for unanswered, or 1 for busy. Well, I hooked the thing up to an AGI script that uses perl and

the caller is out his money anyway when you call any phone and voicemail kicks in, although i think on a payphone they give you a 2 or 3 second window to hang up. Suggest you implement i'm here / i'm away dialplan logic or set the do not disturb button that way when someone calls and the guy is away it hits voicemail right away and the caller can hear this and still have the 2 or 3

Hi David, > The code above is needed as the GHC calling convention redefines what > registers are considered callee save. No one else rummages in to the > original function as all the other calling conventions use the same > set of callee and caller save registers, so GHC is the only one that > needs to differentiate. shouldn't the caller also know what registers are callee

Hi Duncan, Karel, On 14 October 2011 08:22, Duncan Sands <baldrick at free.fr> wrote: > Hi Karel, > >>>> > const unsigned* >>>> > ARMBaseRegisterInfo::getCalleeSavedRegs(const MachineFunction *MF) >>>> const { >>>> > + bool ghcCall = false; >>>> > + >>>> > + if (MF) { >>>> > + const

On 15 October 2011 00:31, Duncan Sands <baldrick at free.fr> wrote: > Hi David, > >> The code above is needed as the GHC calling convention redefines what >> registers are considered callee save. No one else rummages in to the >> original function as all the other calling conventions use the same >> set of callee and caller save registers, so GHC is the only one

I have a rather long dial plan, but it includes support for call waiting. However, the setgroup checkgroup commands don't seem to be working. Can anyone help on this one? Excerpts are below. First exten-vm is dialed and then dial-new. As I understand, priority 1 increments the active channels for the caller and then in "dial-new" priority 8 increments for Arg3, or the Callee

Hi Duncan, Any word on this making 3.0? Cheers, David On 16 October 2011 23:03, David Terei <davidterei at gmail.com> wrote: > On 15 October 2011 00:31, Duncan Sands <baldrick at free.fr> wrote: >> Hi David, >> >>> The code above is needed as the GHC calling convention redefines what >>> registers are considered callee save. No one else rummages in

On Sat, Jun 04, 2005 at 10:09:29AM +0200, Stefan Gofferje arranged a set of bits into the following: > Hi folks, > > there are some 3rd party patches available for chan_sccp which add a > feature and fix a few problems: > > http://www.sineapps.com/news.php?rssid=726 > > sccp_cli.c.diff <http://www.newace.com/asterisk/sccp_cli.c.diff> > Adds support for

I know when you read that subject everyone thinks NAT, but that isn't the case here. Incoming calls get 2 way audio, but outbound calls do not have incoming audio. below is the flow callee --> asterisk --> firewall/router --> provider Callee is firewalled, but not NAT. callee is on the same subnet as the asterisk box. Asterisk box has been completely excluded from the

I have a sipphone.com account, with asterisk set to answer incoming calls, using the following settings (phone number and password omitted) in the Peer Details for the SIP Trunk: allow=ulaw context=from-pstn dtmfmode=rfc2833 fromdomain=proxy01.sipphone.com fromuser=1747xxxxxxx host=proxy01.sipphone.com insecure=very secret=xxxxx type=peer username=1747xxxxxxx The Asterisk machine is

Begin forwarded message: > From: Evan Cheng <evan.cheng at apple.com> > Date: 11 September 2007 19:26:39 GMT+02:00 > To: LLVM Developers Mailing List <llvmdev at cs.uiuc.edu> > Subject: Re: [LLVMdev] RFC: Tail call optimization X86 > Reply-To: LLVM Developers Mailing List <llvmdev at cs.uiuc.edu> > > Hi Arnold, > > Thanks for the patch. Some questions

On Mon, 2008-04-28 at 21:39 -0500, Lane Schwartz wrote: > On Mon, Apr 28, 2008 at 8:31 PM, Gordon Henriksen > <gordonhenriksen at mac.com> wrote: > > On 2008-04-28, at 21:19, Lane Schwartz wrote: > > > > > On Mon, Apr 28, 2008 at 2:13 PM, Gordon Henriksen <gordonhenriksen at mac.com > > >> stack and discover the return address of each call frame.

Hello All, I'am a new Asterisk user, and i have the following question. The following is the Status of all open channels from my Asterisk system, which was received through the Asterisk Manager Interface ((AMI)). ==================================================================== action: Status actionid: 65066874_3# Response: Success ActionID: 65066874_3# Message: Channel status will